Method and device for comparing signals to control transducers and transducer control system

ABSTRACT

A method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones m i  and (p) speakers hp j  for the control of said sound-reproducing systems characterized in that:  
     A: for each speaker hp j ,  
     at least one sound signal S is sent on the speaker hp j ,  
     for each microphone m i , a piece of information hp j m i  is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hp j  and the microphone m i ,  
     B: a reference matrix Q r  is saved, this reference matrix being constituted by all the pieces of reference information hp j m i  obtained following the sending of the sound signal S,  
     C: as soon as a comparison is to be made, the step A is run with a sound signal S′ to obtain current information on a matrix Q,  
     D: the matrices Q and Q r  are compared.

[0001] The invention relates to a method for the automatic comparison of information characterizing reference values and information characterizing current values of sound-reproducing systems of a system of microphones and speakers for the control of the sound-reproducing system.

[0002] The field of the invention is that of the automatic control of the gains, functioning and position of several microphones and several speakers in the context of a system of videoconferencing between participants located at distinct sites that are generally remote sites. The invention can also be applied to the control of microphones and speakers installed in the same room such as a theatre stage, concert hall or cinema hall. It can be used to control the spatialized sound rendition of the scene which provides concordance between visual images and sound. In the videoconferencing context, the invention makes it possible to approach a natural communications situation: when a participant changes position in a remote room during a meeting, the sound follows him in the room in which he is being listened to, with a passage, for example, from one speaker to another as he moves. The microphones and speakers are designated, without distinction, by the term transducers.

[0003] The problem is to detect the changes that occur at the transducers between their installation and the times at which the checks are made.

[0004] An object of the present invention therefore is a method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones m_(i) and (p) speakers hp_(j) for the control of said sound-reproducing systems characterized in that:

[0005] A: for each speaker hp_(j),

[0006] at least one sound signal S is sent on the speaker hp_(j),

[0007] for each microphone m_(i), a piece of information hp_(j)m_(i) is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hp_(j) and the microphone m_(i),

[0008] B: a reference matrix Q_(r) is saved, this reference matrix being constituted by all the pieces of reference information hp_(j)m_(i) obtained following the sending of the sound signal S,

[0009] C: as soon as a comparison is to be made, the step A is run with a sound signal S′ to obtain current information on a matrix Q,

[0010] D: the matrices Q and Q_(r) are compared.

[0011] An object of the invention is also a device for comparing pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of n microphones m_(i) and p speakers hp_(j) for the control of the sound-reproducing system, characterized in that the control system comprises means for the measurement of the pieces of information hp_(j)m_(i) characterizing the sound-reproducing systems comprising a microphone m_(i) and a speaker hp_(j), digital processing means to compare said pieces of information hp_(j)m_(i) and, connected to these digital processing means, means for saving the matrix Q_(r) constituted by all the pieces of information hp_(j)m_(i).

[0012] An object of the invention is also a system for the control of sound-reproducing systems comprising several devices such as those mentioned here above, characterized in that the devices are distributed among several rooms and in that the control system comprises a high bit-rate telecommunications network connecting said rooms and means to centralize the management of the devices.

[0013] Other special features and advantages of the invention shall appear more clearly from the following description given by way of a non-restrictive example, with reference to the appended drawings, of which:

[0014]FIG. 1a) is a diagrammatic view of a videoconferencing room according to the invention,

[0015]FIG. 1b) is a diagrammatic view of the direct paths between speakers and microphones,

[0016]FIGS. 2a) and 2 b) are views of sound-reproducing systems respectively in the case of local processing and when the processing is done in the network,

[0017]FIGS. 3a) and 3 b) respectively show examples of curves representing white noise and USASI noise on the one hand and pink noise and pseudo-random binary sequences on the other hand,

[0018]FIG. 4 shows the impulse response of a microphone following the sending, by a speaker, of a pseudo-random binary sequence,

[0019]FIG. 5 is a diagrammatic view of the configuration of the signal digital processing card,

[0020]FIG. 6 is a diagrammatic view of the system of microphones and speakers distributed among several rooms connected to one another by a multipoint bridge.

[0021] A videoconference is set up between participants distributed among several rooms, a high-bit-rate communications network such as an ATM network being used to convey visual and sound information. A videoconferencing room shown FIG. 1a is provided with a display screen E, several microphones m_(i) and several speakers hp_(j) providing for a spatialized rendition of the audiovisual scene of the remote room or rooms. The speakers may be located, without distinction, all below the screen, all on top or distributed as shown in FIG. 1a, or even in any other arrangement. By way of an indication, the videoconferencing room used for the invention is provided with six microphones and six speakers, the distance between microphones and speakers ranging typically from three to five meters.

[0022] The sound-reproducing systems between the microphones m_(i) and the speakers hp_(j) of a local processing system (shown in FIG. 2a), comprise the microphones m_(i), the microphone preamplifiers am_(i), the analog-digital converters ADC_(i), the digital processing card, the digital-analog converters DAN_(j), the amplifiers of the speakers ahp_(j), the speakers hp_(j) and the room.

[0023] According to another embodiment, the sound-reproducing systems between the microphones m_(i) and the speakers hp_(j) of a remote processing system shown in FIG. 2b), comprise the microphones m_(i), the microphone preamplifiers am_(i), the analog-digital converters ADC_(i), the encoders C_(i), the transportation network R, the decoder D, the digital processing card, the encoder C, the transportation network R, the decoders D_(j), the digital-analog converters DAN_(j), the amplifiers of the speakers ahp_(j), the speakers hp_(j) and the room.

[0024] A routing system A obtained by a multiplexer/demultiplexer also called a switching matrix, which is commercially available, may be inserted if necessary into the sound-reproducing systems between, firstly, the analog-digital converters ADC_(i) and the encoders C_(i) and, secondly, the decoders D_(j) and the analog-digital converters ADC_(j). A remotely controllable system A of this kind makes it possible, at this level of the sound-reproducing system, to route the information characterizing a transducer from one transducer to another.

[0025] Each element of these sound-reproducing systems must be adjusted so as to provide for efficient sound transmission. During the installation of these elements, which is also known as an alignment, the gains, wirings and positions of the transducers of each room are set, and these parameters are stored in a file of a digital processing card of the signal.

[0026] To simplify the matter, the word “transducer” (speaker or microphone respectively) will designate the transducer (the speaker or microphone respectively) and the elements of the sound-reproducing system between the digital processing card and the transducer (speaker or microphone respectively).

[0027] Thereafter, when the videoconference room is used, a week or a month later for example, checks may be made on any modifications that will have occurred in these parameters in order to make the necessary corrections. The transducers may have been moved and in certain cases may have become defective; the room configuration may have been changed; the amplifiers also may have been subjected to high variations over time, possibly caused by the heating of the electronic components. It may be preferred sometimes to act on the transducers in order to compensate for a defect in another element of the sound-reproducing system.

[0028] The term “sound signal” refers to a signal that can be sent by the speakers and detected by the microphones. As indicated in FIGS. 2a) and 2 b), a sound signal S is sent to all the p speakers hp_(j), one after the other at t₁, . . . , t_(j), . . . , t_(p), each in turn, and retrieved at the n microphones m_(i). The reference hp_(j)m_(i) is given to the piece of information characterizing the sound-reproducing system comprising the speaker hp_(j) and the microphone m_(i).

[0029] All these hp_(j)m_(i) pieces of information constitute a matrix with a size n*p, a line of the matrix corresponding to a speaker and a column corresponding to a microphone.

[0030] The first time this matrix is constituted after the alignment, or at another preferred time, it is saved in memory: it is called the reference matrix Q_(r), the elements hp_(j)m_(i) of this matrix being reference values. Thereafter, when a check has to be made on the parameters of these transducers, these steps are reiterated with a signal S′ to obtain current values hp_(j)m_(i) and set up a matrix Q that is compared with the matrix Q_(r).

[0031] In certain cases, it is simpler to choose a signal S′ identical to the signal S, especially when it is sought to compare gains corresponding to the ratio between the energy of the signal sent and the energy of the signal received. In other cases, S is different from S′ and the elements of the matrices Q_(r) and Q to be compared are different in nature. By saving S and S′ and by applying an adequate processing operation to the elements of Q, it is possible to deduce elements comparable to those of Q_(r). With S being known, it is possible to choose a signal S′ that enables, for example, the measurement of the impulse response or the transfer function hp_(j)m_(i) between the transmission point hp_(j) and the reception point m_(i); given S and the characteristics of hp_(j)m_(i), it is possible, from the elements hp_(j)m_(i) of Q, to deduce elements comparable to those of Q_(r) by applying an adequate processing operation (Fourier transform, . . . ).

[0032] It is also possible to set up several matrices Q_(r) by considering several types of signals S and then set up several corresponding matrices Q. If the signal S is, for example, a white noise filtered in different octaves, it is possible to set up a matrix Qr for each octave.

[0033] In general, the elements hp_(j)m_(i) are set up from signals S and S′ considered in the time domain, but it is possible to base the operation on the frequency domain and set up the matrices Q and/or Q_(r) from the spectral responses hp_(j)m_(i) of the microphones m_(i) at a frequency band sent by the speakers hp_(j): whatever the width of the frequency band of the signals S and S′ sent by the speakers hp_(j), only a determined frequency band will be received by the microphones m_(i). It could be a frequency band with a width of about 200 Hz, an octave band or a one-third-octave band. This frequency band will then be made in order to slide to sweep through a spectrum of 0 Hz to 1000 Hz for example.

[0034] During the alignment, the flatness of the spectrum of each transducer is verified, i.e. it is verified that all the frequencies pass through each transducer. If one of them has irregularities, the necessary corrections are made. The microphones sometimes have irregularities related to the table or room effect (to the reflections from the table or room), where the wave reflected by the table or room may be in phase opposition with the direct wave, then giving rise to black regions in the spectral response: the gain of the microphone will then be increased in the corresponding frequency band.

[0035] During subsequent checks, the spectral responses of the transducers by frequency band will be verified. The comparison between the matrices Q and Q_(r) makes it possible, especially, to obtain a piece of information on any movement undergone by the transducers, these transducers being directional and their directivity depending on the frequency. Depending on the results of the comparisons, it is also possible to make a spectral correction to the transducers in order to reduce the coupling between speakers and microphones and cause less deformation in the sound signals sent out by the participants. The exploitation of the results is sometimes more complex than it is when the operation is situated in the time domain.

[0036] The sound signals S and S′ are generally recorded in the internal memory of the signal digital processing card. They may possibly be computed (generated) in this card.

[0037] These sound signals may, for example, be a white noise, a pink noise, an USASI noise, a pseudo-random binary sequence respectively shown in FIGS. 3a) and 3 b) or a sine frequency sweep, an octave-filtered noise or one-third-octave filtered noise, or again another sound signal. Unlike a random noise, a pseudo-random binary sequence is purely deterministic; it is a sequence of 1 and −1 with a length N. The characteristic feature of these sequences is that their correlation function is equal to N for 0 and to −1 for other values. This correlation function is therefore very close to a Dirac distribution.

[0038] The method according to the invention has been carried out with a pink noise sent successively to each of the speakers for one second. Between two sending operations on two consecutive speakers, there is a wait for a certain time (a period of silence) for the next sound signal to start in a state of the sound-reproducing system that is, in principle, a stable state. The invention has been achieved with a two-second period of silence. The elements hp_(j)m_(i) are determined for each hp_(j) at the same instant t of the sound signal. If, for example, hp₁m₁, hp₁m₂, . . . , hp₁m_(n) are determined at t=start of the sound signal+0.9 second, then hp₂m₁, . . . , hp₂m_(n) will be determined at t+3 seconds, hp₃m₁, . . . , hp₃m_(n) at t+6 seconds, etc.

[0039] In adding up and averaging each line and each column of the matrices Q_(r) and Q, possibly after the processing of the elements of a matrix to obtain elements directly comparables to those of the other matrix, a mean value HP_(jQr), HP_(jQ) respectively for each speaker hp_(j) and M_(iQr), M_(iQ) respectively for each microphone m_(i) is obtained. By computing HP_(jQ)/HP_(jQr), we obtain the divergence between the speaker considered and its reference value. Similarly, by computing M_(iQ)/M_(iQr), we obtain the divergence between the microphone itself and its reference value. If, for the speakers as well as the microphones, this divergence is contained in a predetermined range referenced FHP for the speakers and FM for the microphones, then no correction is applied as the difference is tolerable. A threshold of 3 dB is, for example, commonly accepted for a visioconference room. For divergence values outside the predetermined range, a corresponding divergence is applied as a corrective value to the transducer, at the signal digital processing card. As the case may be, the correction could be applied to the gain of the transducer itself. In certain cases, the correction will consist in repositioning the transducer; in other cases, it will not be possible to apply the correction because of a transducer malfunction, and the defective transducer will then be changed.

[0040] The characteristics of the pseudo-random binary sequences make them a preferred signal for the high-precision measurement of the impulse response of a system according to the invention. The use of a pseudo-random binary sequence as a sound signal sent to the speakers hp_(j) therefore enables the measurement of the impulse responses, as a function of time R_(ji), of all the microphones m_(i). Depending on the instant at which the impulse response is considered, each impulse response R_(ji) gives information on the delay, namely, the propagation time between a speaker hp_(j) and a microphone m_(i), the direct wave corresponding to the direct paths between a speaker hp_(j) and microphone m_(i), or again the room effect corresponding to the paths with one or more reflections.

[0041] In FIG. 4, t_(0j) denotes the instant at which the sound signal is sent from a speaker hp_(j), t_(1ji) is the instant at which the microphone m_(i) receives the direct wave and t_(2ji) is the instant at which the room effect starts for the microphone m_(i). It is possible to measure the delays to verify the respective position of the transducers themselves. The matrix Q_(r) is computed by measuring the delays (hp_(j)m_(i))_(Qr) for a first time. The position of the transducers is deduced from these delays by triangulation: if, for example, with the position of hp₁ and hp_(j) being known, the delays (hp₁m₁)_(Qr) and (hp_(j)m₁)_(Qr) are considered, the position of the microphone m₁ when the reference matrix is set up is deduced from this. The same procedure is used for the other microphones. The same reasoning can be applied to determining the position of the speakers from those of the microphones. When the delays (hp_(j)m_(i))_(Q) of the matrix Q are subsequently computed, the transducer that has changed position will subsequently by identified by comparison with the delays of the matrix Q_(r). In certain cases, a correction is applied to the transducer, at the signal digital processing card, to compensate for the change in position. In other cases, the correction will consist in repositioning the transducer itself.

[0042] It is thus possible to evaluate the direct wave resulting from the direct path between the speaker hp_(j) and the microphone m_(i). Each element hp_(j)m_(i) of the matrices Q and Q_(r) then represents the first spike of the impulse response.

[0043] When the evaluation to be made relates to the room effect due to the indirect paths between the speaker hp_(j) and the microphone m_(i), namely the paths of the signals that have undergone various reflections on the walls of the room, on the furniture or on any other obstacle, each element hp_(j)m_(i) of the matrices Q and Q_(r) will represent the part of the impulse response that succeeds the first spike and starts at t_(2ji).

[0044] In one application of the invention, the signal-to-noise ratio of the microphones m_(i) is evaluated by comparing the mean values of the microphones computed from the matrix Q_(r), set up in considering a sound signal S, with the mean values of the microphones computed from the matrix Q set up in considering a signal S′ of silence.

[0045] The signal S may be, especially, a white, rose or USASI noise, or a pseudo-random binary sequence. If the signal S is interspersed with silences, in practice, the signal-to-noise ratio will be measured during a phase of silence.

[0046] It is also possible to remotely process the information characterizing the signals coming from a local room, as a telecommunications or computer network connects the rooms to each other. The information processing comprises especially the measurements, computations, saving operations and corrections to be made. Remote processing can be done by a computer remotely controlling another computer, located in a local room, through the network.

[0047] It is also possible, in the local room, to deal with the case of the remote room or rooms by sending the signals S and S′ through the telecommunications network and retrieving, in the local room, through the network, information characterizing the result of these signals in the remote room or rooms. The same method as described here above is used and, at the level of the signal digital processing card, coefficients are applied to the pieces of information characterizing the transmitted and retrieved signals to have a balanced system.

[0048] An echo phenomenon sometimes occurs: when a participant speaks in a room A, the corresponding sound signal is transmitted to the participants located in a room B by the speakers of this room B, the microphones of this room B taking up the signal coming from these speakers and sending them on to the room A. The speaker of the room A hears himself again with the echo. This echo can be evaluated by measuring the level of the return signal with respect to the level of the signal sent. The control parameters of the echo cancellation or transducer gain variation algorithms are then adjusted.

[0049] It is also possible to comprehensively process the pieces of information hp_(j)m_(i) in the telecommunications network, for example at the level of a multipoint bridge PMP interconnecting several remote rooms Sa, shown in FIG. 6. The signals S and S′ are sent from this bridge to each room Sa through the network and retrieved at this bridge through the network. Precise information on the equipment in each room is not always available. The elements hp_(j)m_(i) are therefore no longer directly linked to the transducers but are linked to the sound-reproducing systems comprising the transmission channels k existing between the bridge PMP and each room Sa. These sound-reproducing systems result, however, for each room, from the sound-reproducing systems internal to these rooms and comprising the speakers hp_(j) and the microphones m_(i). Each room Sa may be connected to the bridge PMP by one or more transmission channels k. For example, two channels could be used for a room to obtain a stereophonic rendition or four could be used to obtain a quadraphonic rendition. If the transmission channels k are numbered 1 to K, then r_(k) for example will designate the sound-reproducing system comprising a transmission channel k transmitting from the room to which it is connected to the bridge PMP and e_(k′) will designate the sound-reproducing system comprising a transmission channel k′ transmitting from the bridge PMP to the room to which it is connected, where k can be equal to k′. The elements hp_(j)m_(i) will then be replaced by r_(k)e_(k′).

[0050] The device according to the invention comprises a signal digital processing card CTN, shown in FIG. 5. This card comprises means Mes for the measurement of the information hp_(j)m_(i), processing means T and file-saving means SF such as an internal memory in which one or more sound signals are recorded. This sound signal may also be computed by the processing means T. The matrix elements hp_(j)m_(i) of the matrix or matrices Q_(r) and, possibly, one of more matrices Q are also saved in the internal memory, along with the parameters of the various elements of each of the sound-reproducing systems obtained during the setting of the room or rooms. The processing means are used to compare elements hp_(j)m_(i) or combinations of these elements belonging to a same matrix Q or to several matrices. They can also be used to compute the corrections to be made to one or more elements of the sound-reproducing system and apply them. They could, for example, correct the gain of a speaker hp_(j) and/or a microphone m_(i). They also enable the generation of a sound signal. These processing means T will be made conventionally by means of a microprocessor P and an associated program memory M comprising a program capable of carrying out the measurements, comparisons, computations and corrections to be made. 

1. Method of comparison between pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of (n) microphones m_(i) and (p) speakers hp_(j) for the control of said sound-reproducing systems characterized in that: A: for each speaker hp_(j), at least one sound signal S is sent on the speaker hp_(j), for each microphone m_(i), a piece of information hp_(j)m_(i) is retrieved, this piece of information characterizing the sound-reproducing system comprising the speaker hp_(j) and the microphone m_(i), B: a reference matrix Q_(r) is saved, this reference matrix being constituted by all the pieces of reference information hp_(j)m_(i) obtained following the sending of the sound signal S, C: as soon as a comparison is to be made, the step A is run with a sound signal S′ to obtain current information on a matrix Q, D: the matrices Q and Q_(r) are compared.
 2. Method according to claim 1 characterized in that, when the pieces of information hp_(j)m_(i) of Q are not directly comparable with the pieces of information of Q_(r), a processing operation, capable of converting them into information directly comparable with the information of the other matrix, is applied to the elements hp_(j)m_(i) from a Q or Q_(r) by choice, before the step D.
 3. Method according to one of the claims 1 or 2 characterized in that the pieces of information hp_(j)m_(i) of Q_(r) and/or of Q are the spectral responses of each sub-system including a speaker hp_(j) and a microphone m_(i).
 4. Method according to claim 3, characterized in that the signals sent by the speakers hp_(j) are transmitted in a frequency band with a determined width, and in that said frequency band is made to slide in order to sweep through the desired spectrum of frequencies.
 5. Method according to one of the claims 1 or 2, characterized in that the pieces of information hp_(j)m_(i) of Q_(r) and/or of Q are the impulse responses of each sub-system including a speaker hp_(j) and a microphone m_(i).
 6. Method according to one of the claims 1 or 2, characterized in that the pieces of information hp_(j)m_(i) of Q_(r) and/or of Q are the transfer functions of each sub-system including a speaker hp_(j) and a microphone m_(i).
 7. Method according to one of the claims 1 or 2, characterized in that the pieces of information hp_(j)m_(i) of Q_(r) and/or of Q are the gains between the microphones m_(i) and the speakers hp_(j) following the signals sent by the speakers hp_(j).
 8. Method according to one of the claims 1 to 7 characterized in that, from the matrices Q and Q_(r), the mean values respectively of the speakers hp_(j), referenced HP_(jQ) and HP_(jQr), are computed by computing: 1/p*Σ_(i) hp_(j)m_(i), and when the value HP_(jQ)/HP_(jQr) is outside a predetermined speaker range FHP, the sound-reproducing system comprising the speaker hp_(j) is corrected by a divergence corresponding to HP_(jQr)/HP_(jQ).
 9. Method according to claim 8, characterized in that, in the sound-reproducing system comprising the speaker hp_(j), the gain of speaker hp_(j) is corrected.
 10. Method according to one of the claims 1 to 7, characterized in that, from the matrices Q and Q_(r), the mean values respectively of the microphones m_(i), referenced M_(iQ) and M_(iQr), are computed by computing: 1/n*Σ_(i) hp_(j)m_(i), and when the value M_(jQ)/M_(jQr) is outside a predetermined microphone range FM, the sound-reproducing system comprising the microphone m_(i) is corrected by a divergence corresponding to M_(iQr)/M_(iQ).
 11. Method according to claim 10, characterized in that, in the sound-reproducing system comprising the microphone m_(i), the gain of the microphone m_(i) is corrected.
 12. Method according to one of the claims 1 to 11, characterized in that the pieces of information hp_(j)m_(i) of the matrices Q and Q_(r) to be compared represent delays between the sending of the sound signal by each speaker hp_(j) and the reception of said sound signal by the microphones m_(i).
 13. Method according to one of the claims 8 to 11, the pieces of information hp_(j)m_(i) of Qr and of Q being the impulse responses of each sub-system including a speaker hp_(j) and a microphone m_(i), characterized in that the pieces of information hp_(j)m_(i) represent the signals that were received by the microphone m_(i) and have traveled the direct path between the speaker hp_(j) and the microphone m_(i).
 14. Method according to one of the claims 8 to 11, the pieces of information hp_(j)m_(i) of Qr and of Q being the impulse responses of each sub-system including a speaker hp_(j) and a microphone m_(i), characterized in that the pieces of information hp_(j)m_(i) represent the signals that were received by the microphone m_(i) and have traveled the paths with one or more reflections between the speaker hp_(j) and the microphone m_(i).
 15. Method according to one of the claims 1 to 14, characterized in that the signal S′ is a silence signal and in that, from the matrices Q and Q_(r), the mean values of the microphones m_(i), referenced M_(iQ) and M_(iQr), are computed respectively in computing: 1/n*Σ_(j) hp_(j)m_(i), to thus obtain the signal-to-noise ratio M_(iQr)/M_(iQ) of the microphones m_(i).
 16. Method according to one of the above claims, characterized in that the pieces of information hp_(j)m_(i) are processed remotely through a telecommunications or computer network.
 17. Method according to one of the claims 1 to 15 characterized in that the signals S and S′ come from a remote room connected to a local room through a telecommunications network and wherein the pieces of information hp_(j)m_(i) are processed in the local room.
 18. Method according to one of the claims 1 to 11, characterized in that the pieces of information hp_(j)m_(i) of Q_(r) and Q to be compared represent the echo and in that the signals S and S′ come from a remote room connected to a local room through a telecommunications network.
 19. Method for the comparison of information characterizing sound-reproducing systems, at a point of a telecommunications network connecting remote rooms equipped with a system of n microphones m_(i) and p speakers hp_(j), each room being connected to the point of the network by one or more transmission channels k, characterized in that the sound-reproducing systems r_(k)e_(k′) comprise a transmission channel k and a transmission channel k′ resulting from one or more sound-reproducing systems hp_(j)m_(i) internal to each room processed according tone of the claims 1 to 18, and in that the elements of the matrices Q and Q_(r) represent pieces of information characterizing the sound-reproducing systems r_(k)e_(k′).
 20. Method according to one of the above claims, characterized in that the signals S and/or S′ are constituted by a white noise or a pink noise or an USASI noise or a pseudo-random binary sequence.
 21. Method according to one of the claims 1 to 20, characterized in that it consists in using the same signal S to obtain the matrices Q_(r) and Q.
 22. Device for comparing pieces of information characterizing reference values and pieces of information characterizing current values of sound-reproducing systems of a system of n microphones m_(i) and p speakers hp_(j) for the control of the sound-reproducing system, characterized in that the control system comprises means for the measurement of the pieces of information hp_(j)m_(i) characterizing the sound-reproducing systems comprising a microphone m_(i) and a speaker hp_(j), digital processing means to compare said pieces of information hp_(j)m_(i) and, connected to these digital processing means, means for saving the matrix Q_(r) constituted by all the pieces of information hp_(j)m_(i).
 23. Device according to claim 22, characterized in that the digital processing means comprise means for the comparison of the matrix Q_(r) and a matrix Q constituted by pieces of information hp_(j)m_(i) characterizing the current values.
 24. Device according to claim 22 or 23, characterized in that the digital processing means comprise means to generate sound signals (S) on the speakers hp_(j).
 25. Device according to claim 24, characterized in that the sound signals (S) may be a white noise or a pink noise or an USASI noise or a pseudo-random binary sequence.
 26. Device according to one of the claims 22 to 25, characterized in that the processing means comprise means to correct the sound-reproducing system comprising a speaker hp_(j) and a microphone m_(i).
 27. Device according to claim 26, characterized in that, in the sound-reproducing system comprising the speaker hp_(j), the gain of the speaker hp_(j) is corrected.
 28. Device according to claim 26, characterized in that, in the sound-reproducing system comprising the microphone m_(i), the gain of the microphone m_(i) is corrected.
 29. System for the control of sound-reproducing systems comprising several devices according to one of the claims 22 to 28, characterized in that the devices are distributed among several rooms and in that the control system comprises a high bit-rate telecommunications network connecting said rooms and means to centralize the management of the devices.
 30. System according to claim 29, characterized in that the means to centralize the management of the devices are located at a point of the telecommunications network connecting the remote rooms, each room being connected to the point of the network by one or more transmission channels k, in that the sound-reproducing systems r_(k)e_(k′) comprising a transmission channel k and a transmission channel k′ result from one or more sound-reproducing systems hp_(j)m_(i) internal to each room and in that it comprises means to correct the sound-reproducing systems r_(k)e_(k′). 